Introduce RtpStreamReceiver and RtpStreamReceiverControllerInterface.

webrtc/call/rtp_stream_receiver_controller.h

+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef WEBRTC_CALL_RTP_STREAM_RECEIVER_CONTROLLER_H_
+#define WEBRTC_CALL_RTP_STREAM_RECEIVER_CONTROLLER_H_
+
+#include <memory>

[the sun, 2017/06/16 13:37:39]

Not needed (unique_ptr should be included in
rtp_stream_receiver_controller_interface.h already)

[nisse-webrtc, 2017/06/16 14:33:26]

I agree, but presubmit doesn't... If I delete that line, I get

.../rtp_stream_receiver_controller.cc:31:  Add #include <memory> for
unique_ptr<>  [build/include_what_you_use] [4]
.../rtp_stream_receiver_controller.h:32:  Add #include <memory> for unique_ptr<>
 [build/include_what_you_use] [4]

+
+#include "webrtc/base/array_view.h"

[the sun, 2017/06/16 13:37:39]

Not needed

[nisse-webrtc, 2017/06/16 14:33:26]

Done.

+#include "webrtc/base/criticalsection.h"
+

[the sun, 2017/06/16 13:37:39]

dd

[nisse-webrtc, 2017/06/16 14:33:26]

I'm not following you.

[the sun, 2017/06/16 15:00:13]

Sorry, vim-speak for delete a line. It's the only thing from my 5 weeks with vim
that stuck. :)

[nisse-webrtc, 2017/06/19 07:21:01]

Done. I would have understood C-k (short for M-x kill-line RET) better ;-)

+#include "webrtc/call/rtp_demuxer.h"
+#include "webrtc/call/rtp_stream_receiver_controller_interface.h"
+
+namespace webrtc {
+
+class ReceiveSideCongestionController;

[the sun, 2017/06/16 13:37:39]

not used

[nisse-webrtc, 2017/06/16 14:33:26]

Done.

+class RtpPacketReceived;
+
+// This class represents the RTP receive parsing and demuxing, for a
+// single RTP session.
+// TODO(nisse): Add RTCP processing, we should aim to terminate RTCP
+// and not leave any RTCP processing to individual receive streams.
+// TODO(nisse): Extract per-packet processing, including parsing and
+// demuxing, into a separate class.
+class RtpStreamReceiverController
+    : public RtpStreamReceiverControllerInterface {
+ public:
+  // Implements RtpStreamReceiverControllerInterface.
+  std::unique_ptr<RtpStreamReceiverInterface> CreateReceiver(
+      uint32_t ssrc,
+      RtpPacketSinkInterface* sink) override;
+
+  // Thread-safe wrappers for the corresponding RtpDemuxer methods.
+  void AddSink(uint32_t ssrc, RtpPacketSinkInterface* sink) override;
+  size_t RemoveSink(const RtpPacketSinkInterface* sink) override;
+
+  // TODO(nisse): Not yet responsible for parsing.
+  bool OnRtpPacket(const RtpPacketReceived& packet);
+
+ private:
+  class Receiver : public RtpStreamReceiverInterface {

[the sun, 2017/06/16 13:37:39]

Is there any reason to not put this in an anonymous namespace in the .cc?

[nisse-webrtc, 2017/06/16 14:33:26]

I think it's tied to this class in such a way that it makes sense for it to be
declared here. And we get the option of making the RemoveSink method private
(only use is in the Receiver destructor), and we might want to add more private
methods later.

Is it possibly to forward declare it as an inner class here (for namespace
reasons), but move the rest of the declaration to the implementation file? 

I don't have a strong opinion either way, and I must admit that I don't really
know what a private class implies for C++ access rules, so I might have
misunderstood. So I'll do whatever you prefer.

[the sun, 2017/06/16 15:00:13]

Ah, didn't see that. Good call.

+   public:
+    Receiver(RtpStreamReceiverController* controller,
+             uint32_t ssrc,
+             RtpPacketSinkInterface* sink);
+
+    ~Receiver() override;
+
+   private:
+    RtpStreamReceiverController* const controller_;
+    RtpPacketSinkInterface* const sink_;
+  };
+
+  rtc::CriticalSection lock_;
+  RtpDemuxer demuxer_ GUARDED_BY(&lock_);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_CALL_RTP_STREAM_RECEIVER_CONTROLLER_H_